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Author SHA1 Message Date
Sebastian Jansson
c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation."" 
This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
 2020-01-29 18:42:34 +00:00
Bjorn A Mellem
0cda7b832a Allow non-identical datagram transport parameters. 
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport.  This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).

This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.

Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
 2020-01-29 18:14:24 +00:00
Mirko Bonadei
4356490b7b Revert "Reland "Only include overhead if using send side bandwidth estimation."" 
This reverts commit 086055d0fd9b9b9efe8bcf85884324a019e9bd33.

Reason for revert: Causes some perf regressions.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
> 
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
> 
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> > 
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
> 
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11298
Change-Id: Id38de92ac25a1ce9a1360f0e37f65747d4cfb31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30411}
 2020-01-29 16:38:57 +00:00
Jerome Jiang
740ed473dc Add 444 support for vp9 decoder wrapper. 
Chromting is trying vp9 444 to have better color. This fix is needed to decode 444 properly.

Bug: webrtc:11326
Change-Id: I4498930591d8876af9f6b7238a8c9fe450ecbfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166220
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30410}
 2020-01-29 16:32:10 +00:00
Mirko Bonadei
7affd9bcbb Revert "Adds trial to use correct overhead calculation in pacer." 
This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7.

Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict.

Original change's description:
> Adds trial to use correct overhead calculation in pacer.
> 
> Bug: webrtc:9883
> Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30399}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30409}
 2020-01-29 15:29:04 +00:00
Minyue Li
99d6d8115b Adding absolute capture timestamp to AudioTrackSinkInterface. 
Bug: webrtc:10739
Change-Id: I8c134cbe82452ac71625cd0c810c783a73f17822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167532
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30408}
 2020-01-29 13:46:28 +00:00
Danil Chapovalov
26b4cb3fc5 Detach RtpFrameReferenceFinder from RtpGenericFrameDescriptor 
To allow to use the RtpFrameReferenceFinder with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: Ib60a505a714993862a008300aa64d0bb835c3377
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167361
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30407}
 2020-01-29 12:36:10 +00:00
Evan Shrubsole
73a5e916a9 Remove task_queue dependency for QualityScaler 
This allows for the possiblity to move the QualityScaler
out of the VideoStreamEncoder in the future.


Bug: webrtc:11222
Change-Id: I1d563cf08791e27ff5065ce90bcb150a7974d868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167534
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30406}
 2020-01-29 12:14:10 +00:00
Ivo Creusen
182c2b8334 Expose run function to NetEqSimulator 
Bug: webrtc:11005
Change-Id: I84f01536b40ba17e66877cdced194e05b882b5c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167537
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30405}
 2020-01-29 11:55:05 +00:00
Danil Chapovalov
97ffbefdab Pass and store PacketBuffer::Packet by unique_ptr 
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.

Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
 2020-01-29 11:48:55 +00:00
Sebastian Jansson
d7fade5738 Makes all units and operations constexpr 
Since RTC_DCHECK was made constexpr compatible, we can now
make the unit classes fully constexpr.

Bug: webrtc:9883
Change-Id: I18973c2f318449869cf0bd45699c41be53fba806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30403}
 2020-01-29 10:57:54 +00:00
Rikard Lundmark
48be482d73 Fix spelling. 
Bug: None
Change-Id: Id281fe3d58bd5a8651b299b426353524085dd876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167536
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rikard Lundmark <lundmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30402}
 2020-01-29 10:50:14 +00:00
Yves Gerey
071d025929 Activate event tracing for unit tests. For good! 
The --trace_event=file.json option allows to log events,
for further inspection in chromium event viewer.

Previous handling of this option was broken,
closing the logger before the tests were even run.

Bug: webrtc:10926
Change-Id: I9123d12666b5f254feeaef685def96eb8ba1c7f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30401}
 2020-01-29 10:11:34 +00:00
philipel
52c62df2ed Don't condition the time_controller target on rtc_include_tests. 
Bug: none
Change-Id: Ifb3f811c71a778a447c41593902c417614ae9824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167723
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30400}
 2020-01-29 09:59:34 +00:00
Sebastian Jansson
71a77c4b3b Adds trial to use correct overhead calculation in pacer. 
Bug: webrtc:9883
Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30399}
 2020-01-29 09:39:40 +00:00
chromium-webrtc-autoroll
262cf691b1 Roll chromium_revision fa85f826d0..dd5a54c29b (736081:736224) 
Change log: fa85f826d0..dd5a54c29b
Full diff: fa85f826d0..dd5a54c29b

Changed dependencies
* src/base: 72eb032b0e..ce806f00e6
* src/build: 5dc5e02620..2f17606c25
* src/ios: 0de24610a1..31829ea7dd
* src/testing: b83b82bf65..403d2930e5
* src/third_party: aa87e20519..18f4ad54fc
* src/third_party/depot_tools: 9d635962bc..ea8b58b970
* src/tools: 03d0d67bf3..39d70a2950
DEPS diff: fa85f826d0..dd5a54c29b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I5ab5844dcfc879598a31aeb24f6b8687ec497fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167780
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30398}
 2020-01-29 04:51:45 +00:00
chromium-webrtc-autoroll
5d3173be30 Roll chromium_revision 5146474c0d..fa85f826d0 (735951:736081) 
Change log: 5146474c0d..fa85f826d0
Full diff: 5146474c0d..fa85f826d0

Changed dependencies
* src/base: 634449a5bf..72eb032b0e
* src/build: 9f84364df2..5dc5e02620
* src/buildtools: 48cce924d6..afc5b798c7
* src/buildtools/linux64: git_revision:83dad00afb232d7235dd70dff1ee90292d72a01e..git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65
* src/buildtools/mac: git_revision:83dad00afb232d7235dd70dff1ee90292d72a01e..git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65
* src/buildtools/win: git_revision:83dad00afb232d7235dd70dff1ee90292d72a01e..git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65
* src/ios: 2a0039e931..0de24610a1
* src/testing: b0b0eefbd9..b83b82bf65
* src/third_party: f6957794bd..aa87e20519
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/315382afa6..2e0a0cb9ad
* src/tools: 1a5d6fe796..03d0d67bf3
DEPS diff: 5146474c0d..fa85f826d0/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9064ac861d5060dda4a3bf67306ae0cb295c3476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167760
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30397}
 2020-01-28 22:59:08 +00:00
Danil Chapovalov
b6bf0b2546 Pass picture_id from generic packetizer through codec-specific field 
To free up RtpVideoHeader::generic field for codec agnostic details
from an rtp header extension.

Bug: webrtc:10342
Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30396}
 2020-01-28 19:26:28 +00:00
Steve Anton
f417238217 Remove iceRegatherIntervalRange 
This was an ICE configuration experiment added a couple years ago that did not end up being used.

Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
 2020-01-28 19:16:18 +00:00
chromium-webrtc-autoroll
ed9a401f27 Roll chromium_revision 0168397940..5146474c0d (735581:735951) 
Change log: 0168397940..5146474c0d
Full diff: 0168397940..5146474c0d

Changed dependencies
* src/base: e65cf566c2..634449a5bf
* src/build: e996a848b0..9f84364df2
* src/ios: d2debbb1b0..2a0039e931
* src/testing: c9b319f108..b0b0eefbd9
* src/third_party: 9d6f0a885b..f6957794bd
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3906f655f4..315382afa6
* src/third_party/depot_tools: f437869d41..9d635962bc
* src/tools: d0ce076f32..1a5d6fe796
DEPS diff: 0168397940..5146474c0d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I0f717a804f297855f24f4ceef4590dbd50b2c3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167703
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30394}
 2020-01-28 18:49:48 +00:00
Philipp Hancke
8c52e8a2ef remove mention of prebuilt libraries from docs/ 
deprecated per M80 release notes:
  https://groups.google.com/forum/?#!msg/discuss-webrtc/Ozvbd0p7Q1Y/M4WN2cRKCwAJ

BUG=none

Change-Id: If08537d696baee67626f20996e4f5de261ebee76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167535
Reviewed-by: Anatoli Davidson <anatolid@webrtc.org>
Commit-Queue: Anatoli Davidson <anatolid@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30393}
 2020-01-28 16:08:44 +00:00
Per Åhgren
260c788d77 AEC3: Added multi-channel support for the capture delay functionality 
This CL adds the missing support for multi-channel in the code that
provides an optional and configurable delay to be added to the
microphone signal.

The CL also makes the creation of the delay object conditional on the
need for that support (this is important since this adds a significant
heap memory footprint)

Bug: webrtc:11314,chromium:1045910
Change-Id: I92d577e31af830945fe9d5ca2032000aad4266be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167525
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30392}
 2020-01-28 15:39:26 +00:00
Henrik Boström
4a5dab00ae [Stats] Include fecPackets[Reeceived/Discarded] in Members() 
This refers to modern getStats() only. The metrics has been implemented
for a while in C++ but was accidentally not included in the Members()
list, meaning they were not exposed in lists (including exposure in
Chrome/JavaScript).

The Chromium whitelist already include them.

TBR=hta@webrtc.org

Bug: webrtc:11317
Change-Id: I0c3ee9c552975fc37db2d87196c66e662c994aed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167530
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30391}
 2020-01-28 11:22:09 +00:00
Sebastian Jansson
086055d0fd Reland "Only include overhead if using send side bandwidth estimation." 
This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
 2020-01-28 10:36:39 +00:00
Jonas Oreland
7a284e1614 TCPConnection: Defer FailAndPrune by signaling to self 
What steps will reproduce the problem?
1. Connect a TCPPort, creating a TCPConnection
2. Disconnect the interface (e.g turn it off in android)
3. Send Ping on the TCPConnection

Crash.
The TCPConnection calls FailAndPrune when it fails to reconnect
the TCPConnection. FailAndPrune which removes the StunRequests.
When this is called from the Ping() code,
that will still access the StunRequest after the call to the Connection.

Solution: Instead of calling FailAndPrune deep down in the Ping()-stack
post a message to self to do this with a "clean" stack instead.

BUG: webrtc:11315
Change-Id: Id328b1b7c92311fa5b9adbfd2eb1dd14bf19805d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167522
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30389}
 2020-01-28 09:50:06 +00:00
Sam Zackrisson
6136fdb287 Whitespace change 
Bug: None
TBR: mbonadei@webrtc.org
No-Try: True
Change-Id: I0a93a68610bb4837be4fff6550675759ee1f59b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167529
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30388}
 2020-01-28 07:53:15 +00:00
Steve Anton
8a6f9a03f0 Export IceParameters::Parse for use in Chrome 
Bug: chromium:1044521
Change-Id: I7c6fb0ba5ac918858ed65f9fe503c4de6f6acce5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167683
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30387}
 2020-01-28 01:46:04 +00:00
chromium-webrtc-autoroll
11b66cf110 Roll chromium_revision 5a8e8ca513..0168397940 (735421:735581) 
Change log: 5a8e8ca513..0168397940
Full diff: 5a8e8ca513..0168397940

Changed dependencies
* src/base: 939b5844e6..e65cf566c2
* src/build: 83cc2ebf32..e996a848b0
* src/ios: b9ca807521..d2debbb1b0
* src/testing: f37f2d115c..c9b319f108
* src/third_party: 87b241ffe2..9d6f0a885b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d93fde1cd5..3906f655f4
* src/third_party/depot_tools: 0aa48cc1de..f437869d41
* src/third_party/libvpx/source/libvpx: 7763c888e0..4254ecaa07
* src/tools: 5d93d4e276..d0ce076f32
DEPS diff: 5a8e8ca513..0168397940/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org, jianj@chromium.org,
BUG=None

Change-Id: Ife5209fac61ef0b698ca0a54dc5cae47be711208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167681
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30386}
 2020-01-27 23:01:12 +00:00
Sebastian Jansson
17a6381c1c Adds fake video codec mode to PeerScenarioClient 
This improves execution speed by skipping the encoding step.

Bug: webrtc:10365
Change-Id: I6aef1376c157d859f05f4a44f881d1c60f353067
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167082
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30385}
 2020-01-27 18:07:45 +00:00
chromium-webrtc-autoroll
2e4f440fde Roll chromium_revision 08a3245b28..5a8e8ca513 (735303:735421) 
Change log: 08a3245b28..5a8e8ca513
Full diff: 08a3245b28..5a8e8ca513

Changed dependencies
* src/base: b97b755f7b..939b5844e6
* src/build: cbcd766952..83cc2ebf32
* src/ios: e7e769c1c8..b9ca807521
* src/third_party: 4ec85c3ce8..87b241ffe2
* src/third_party/harfbuzz-ng/src: 82545c5e2b..5440313924
* src/tools: 06243dfa4f..5d93d4e276
DEPS diff: 08a3245b28..5a8e8ca513/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: If691c44d238ca18192b7b1b165d2dbc58ab8fbd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167600
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30384}
 2020-01-27 16:44:10 +00:00
Sebastian Jansson
c709412c76 Revert "Only include overhead if using send side bandwidth estimation." 
This reverts commit 8c79c6e1af354c526497082c79ccbe12af03a33e.

Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
 2020-01-27 15:09:49 +00:00
Sebastian Jansson
8c79c6e1af Only include overhead if using send side bandwidth estimation. 
Bug: webrtc:11298
Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30382}
 2020-01-27 14:19:54 +00:00
Henrik Boström
ad515a255b [Overuse] Move GetCpuOveruseOptions() to adaption module. 
This removes the last remaining explicit reference from
OveruseFrameDetectorResourceAdaptationModule to
VideoStreamEncoder.

VideoStreamEncoder's call to SetEncoderSettings() inside
ReconfigureEncoder() is moved a few lines down - it was discovered that
during these lines the EncoderInfo config could get modified in
response to InitEncode() - so this fixes a potential bug.

Bug: webrtc:11222
Change-Id: I9746f28a4df8e631e297669c10636bf17b39acec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167363
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30381}
 2020-01-27 13:51:40 +00:00
Minyue Li
ff0e4dbd1f Reland "Send absolute capture time through audio coding module." 
This is a reland of 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a

Original change's description:
> Send absolute capture time through audio coding module.
>
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

Bug: webrtc:10739
Change-Id: I10086d239ad3f1efb8485098bf3b0ad23110962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167213
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30380}
 2020-01-27 13:18:27 +00:00
Jonas Olsson
6c9bc396e9 Cleanup log formatting in modules/audio_processing 
Bug: None
Change-Id: I47177530d8a85d7b2f143081de71f5a3bf8ec354
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166041
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30379}
 2020-01-27 09:42:56 +00:00
Patrik Höglund
9c0a83ea4d Remove strip_absolute_paths_from_debug_symbols from mb. 
Getting failures on more_configs bots, e.g.
https://ci.chromium.org/p/webrtc/builders/try/android_arm_more_configs/16421

Appears strip_absolute_paths_from_debug_symbols has changed behavior.
It's now forced on if the platform does it by default and not
configurable. Therefore our bots can't set it explictly on most
platforms.

Bug: None
Change-Id: I112dbb18f9d19ba3dc645a6ae640098afac3c408
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167520
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30378}
 2020-01-27 09:17:46 +00:00
chromium-webrtc-autoroll
c99afa0628 Roll chromium_revision 812b6f8943..08a3245b28 (735202:735303) 
Change log: 812b6f8943..08a3245b28
Full diff: 812b6f8943..08a3245b28

Changed dependencies
* src/build: 00863bc5a5..cbcd766952
* src/ios: 71cb14cc3a..e7e769c1c8
* src/third_party: 7ce74865ec..4ec85c3ce8
* src/third_party/depot_tools: bf4e7a722b..0aa48cc1de
* src/tools: cb743f7070..06243dfa4f
DEPS diff: 812b6f8943..08a3245b28/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Iae7818ed3ef49bffe6b27690827252ba21a34294
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167473
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30377}
 2020-01-26 22:19:32 +00:00
chromium-webrtc-autoroll
4c4735bfcc Roll chromium_revision c04519686a..812b6f8943 (734489:735202) 
Change log: c04519686a..812b6f8943
Full diff: c04519686a..812b6f8943

Changed dependencies
* src/base: e1626e708d..b97b755f7b
* src/build: 08c5083ab8..00863bc5a5
* src/ios: 027fc768d4..71cb14cc3a
* src/testing: 8d8f6f0831..f37f2d115c
* src/third_party: 8966ce7713..7ce74865ec
* src/third_party/android_build_tools/bundletool: VVve-571EEzx-giwEOU0vCrIe9D9a_TjP6ka1GgeVSUC..P0-ZY8wc-hAu5TZYFH7bId8H9Ucy7mNGCg1IPzXuZpEC
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotation: version:2.3.3-cr0..version:2.3.4-cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotations: version:2.3.3-cr0..version:2.3.4-cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_check_api: version:2.3.3-cr0..version:2.3.4-cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_core: version:2.3.3-cr0..version:2.3.4-cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_type_annotations: version:2.3.3-cr0..version:2.3.4-cr0
* src/third_party/android_deps/libs/org_checkerframework_checker_qual: version:2.5.3-cr0..version:3.0.0-cr0
* src/third_party/android_deps/libs/org_checkerframework_dataflow: version:2.5.3-cr0..version:3.0.0-cr0
* src/third_party/android_deps/libs/org_checkerframework_javacutil: version:2.5.3-cr0..version:3.0.0-cr0
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/89730072b8..1cc95ac07c
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/68125d9096..d93fde1cd5
* src/third_party/depot_tools: bdd89366d3..bf4e7a722b
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/7bde79cc27..f70e2f1641
* src/tools: 1a4aa4db04..cb743f7070
Added dependencies
* src/third_party/android_deps/libs/com_github_ben_manes_caffeine_caffeine
* src/third_party/android_deps/libs/org_plumelib_require_javadoc
* src/third_party/android_deps/libs/org_plumelib_reflection_util
* src/third_party/android_deps/libs/org_plumelib_plume_util
Removed dependency
* src/third_party/android_deps/libs/com_github_stephenc_jcip_jcip_annotations
DEPS diff: c04519686a..812b6f8943/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia764fcff235a34e69432aadabe058689e4a308f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167423
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30376}
 2020-01-25 02:37:26 +00:00
Steve Anton
71ff073698 Validate ICE ufrag/pwd according to the spec 
https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-39#section-5.4

Bug: chromium:1044521
Change-Id: Ia95718437dfc270b52cdf822e861a3da7cbbab76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167281
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30375}
 2020-01-25 01:38:50 +00:00
Jamie Walch
f3886aea86 Include cursor rects in updated_region. 
DesktopAndCursorComposer adds the cursor image to the desktop, but does
not change the updated_region, so it generally doesn't encode correctly
unless the mouse is moving over a region that is changing. This CL
extends the updated region to include the union of the old and new
cursor rects, with an optimization for the case where the cursor has
neither moved nor changed.

Bug: chromium:1043325
Change-Id: I52076c96528820833fda6aa95f5b1fbc0f613909
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166545
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30374}
 2020-01-24 20:16:58 +00:00
Johannes Kron
a104ceb0ce Revert "Reland "Reland "Distinguish between send and receive codecs""" 
This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.

Reason for revert: Breaks perf test on iOS.

Original change's description:
> Reland "Reland "Distinguish between send and receive codecs""
> 
> This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> 
> Reason for revert: Flaky test in Chromium fixed.
> 
> Original change's description:
> > Revert "Reland "Distinguish between send and receive codecs""
> > 
> > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > 
> > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > 
> > Original change's description:
> > > Reland "Distinguish between send and receive codecs"
> > > 
> > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > 
> > > Reason for revert: Fixed negotiation of send-only clients.
> > > 
> > > Original change's description:
> > > > Revert "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > >
> > > > Reason for revert: breaks negotiation with send-only clients
> > > >
> > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive codecs
> > > > >
> > > > > Even though send and receive codecs may be the same, they might have
> > > > > different support in HW. Distinguish between send and receive codecs
> > > > > to be able to keep track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > 
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > 
> > > 
> > > Bug: chromium:1029737
> > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30348}
> > 
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30360}
> 
> TBR=steveanton@webrtc.org,kron@webrtc.org
> 
> Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30367}

TBR=steveanton@webrtc.org,kron@webrtc.org

Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30373}
 2020-01-24 16:44:17 +00:00
Sam Zackrisson
b039c30157 Reland "Change log level of AEC3 buffer info to VERBOSE" 
This is a reland of 48148dc840f66c5f6adc5e2ba01c15104e0a9bab

Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
>
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
>
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}

Bug: webrtc:11278, webrtc:11295
Change-Id: I8e6f11457e283c83cae5581adcacdc4d3b5431bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30372}
 2020-01-24 12:58:08 +00:00
Artem Titov
1e02339ea6 Add ability to set custom adapter type on emulated endpoint 
Bug: webrtc:10138
Change-Id: I2f53b42a2c377c9c0c9d36b61eb1c6ce96da480a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167209
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30371}
 2020-01-24 12:53:07 +00:00
Sam Zackrisson
b18c4eb0a9 Add parameterization for three multi channel AEC3 unit tests 
Bug: webrtc:11295
Change-Id: I478aa02908c494cf9609db00021438a59a132b66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167202
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30370}
 2020-01-24 12:26:46 +00:00
Danil Chapovalov
159c414ff8 Detach LossNotificationController from RtpGenericFrameDescriptor 
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
 2020-01-24 11:53:28 +00:00
Ivo Creusen
88636c6dac Improvements for NetEqControllers 
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.

Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
 2020-01-24 11:39:52 +00:00
Johannes Kron
9bac68c0cc Reland "Reland "Distinguish between send and receive codecs"" 
This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.

Reason for revert: Flaky test in Chromium fixed.

Original change's description:
> Revert "Reland "Distinguish between send and receive codecs""
> 
> This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> 
> Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> 
> Original change's description:
> > Reland "Distinguish between send and receive codecs"
> > 
> > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > 
> > Reason for revert: Fixed negotiation of send-only clients.
> > 
> > Original change's description:
> > > Revert "Distinguish between send and receive codecs"
> > >
> > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > >
> > > Reason for revert: breaks negotiation with send-only clients
> > >
> > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > >
> > > Original change's description:
> > > > Distinguish between send and receive codecs
> > > >
> > > > Even though send and receive codecs may be the same, they might have
> > > > different support in HW. Distinguish between send and receive codecs
> > > > to be able to keep track of which codecs have HW support.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30292}
> > 
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> > 
> > 
> > Bug: chromium:1029737
> > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30348}
> 
> TBR=steveanton@webrtc.org,kron@webrtc.org
> 
> Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30360}

TBR=steveanton@webrtc.org,kron@webrtc.org

Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30367}
 2020-01-23 23:02:59 +00:00
Steve Anton
760fd52494 Replace MockAudioDeviceModule mock refcounting with real refcounting 
Bug: webrtc:11308
Change-Id: Ic55ec2c4b45f8fc709fe1348556bdeea6202e7a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166580
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30366}
 2020-01-23 19:04:58 +00:00
chromium-webrtc-autoroll
40899b2797 Roll chromium_revision 487ee81fa3..c04519686a (734357:734489) 
Change log: 487ee81fa3..c04519686a
Full diff: 487ee81fa3..c04519686a

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6b345995a8..68125d9096
DEPS diff: 487ee81fa3..c04519686a/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ib06e84108ad18eb8093db654eb38f090ab7673d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167221
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30365}
 2020-01-23 16:40:14 +00:00
Minyue Li
4175914f41 Revert "Send absolute capture time through audio coding module." 
This reverts commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a.

Reason for revert: failing upstream tests

Original change's description:
> Send absolute capture time through audio coding module.
> 
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com

Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
 2020-01-23 16:21:06 +00:00